Grandstream VoIP Phones

Who else on the list is using them, particularly in a hosted environment?
We’ve just decided to transition to them as our primary recommendation instead of the Cisco SPA series. We did it because of the value and feature set, like having an inexpensive phone with a small BLF, which a lot of customers asked for. I’m wondering if others have tips they’ve learned along the way, or any advice they want to offer. Also anyone using the advanced features like the browser for anything useful?

For those who haven’t tried them, or who like us, didn’t like their older models, take another look. We have been surprised at the value they give us. The prices are low, but the functionality and quality are high. They aren’t Polycom 600s to be sure, but they are nice phones that have a huge set of features for a great price. Customers are liking them a lot.

Has anyone used the new DECT phone? We currently use the Panasonic DECT
phones but they are a nightmare to configure.

If anyone wants to get in touch with them, our Grandstream contact is Dennis Ryan, dryan@grandstream.com .

12 Responses to “Grandstream VoIP Phones”

  1. Vladimir Mikhelson said:

    Aug 17, 12 at 11:09 am

    Carlos,

    I am waiting for my Grandstreams to arrive too.

    Similar reasons. Great feature set, reasonable price.

    My primary interest is security. Grandstream claims their intermediate and higher-end models support TLS and SRTP. I am really tired of trying to make Cisco phones to communicate securely with Asterisk. Cisco has a great security model but one has to have their provisioning server for it to function.

    Regards, Vladimir

  2. Carlos Alvarez said:

    Aug 17, 12 at 11:36 am

    We’ve never had customers ask for this, but if doing so is fairly easy we would look at it as just another feature we push. Do let me know how it works out for you.

  3. Jeff LaCoursiere said:

    Aug 17, 12 at 11:58 am

    For what its worth, Grandstream has burned way too many bridges. We are now happy resellers of Yealink, and as far as security goes, the ability of the phone to do OpenVPN natively is now indispensable for us.

    Cheers,

    j

  4. Bryant Zimmerman said:

    Aug 17, 12 at 12:21 pm

    We are very happy with most Grandstream products we work very closely with their engineering and support guys they are very good about getting us new features and bug fixes, and yes like most VOIP hardware they do have lot’s of bugs but they actually fix most of them.

    The only current products I don’t recommend from them right now are the GXV-3175 (Slow processor and very buggy firmware), GXV-3000 (Has a handset mic overdrive issue. Have to add a resistor to the inside of the handset). I am holding on the new Dect DP715 for the next firmware rev (it is a good product at a fair price but several firmware bugs are show stoppers. We are awaiting beta fixes sometime next week and then we will start shipping them) Hope this helps.

    zktech

  5. Vladimir Mikhelson said:

    Aug 31, 12 at 8:07 pm

    Carlos,

    So far the experience with DP715 is extremely negative.

    It all starts with the WEB interface which is only served on port 80, no https, period. There is no login name, just password.

    The phone worked as expected with insecure SIP and RTP. As I started playing with security the phone started acting up. It randomly took calls, then stopped. It placed calls, then stopped.

    Following is a sample of a corrupted SIP message Asterisk receives from DP715 (pay attention to Call-ID: 477744485-5061-8@BHC.BH.BDH.HB):

    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0
    200 OK
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via:
    SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rportP61
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From:
    ;tag=as50c4dc59
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To:
    ;tagC6538044
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID:
    477744485-5061-8@BHC.BH.BDH.HB
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact:

    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]:
    Supported: replaces, path, timer, eventlist
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]:
    User-Agent: Grandstream DP715 1.0.0.5
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow:
    INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
    [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]:
    Content-Length: 0

    According to RFC 3261, “Call-ID contains a globally unique identifier for this call, generated by the combination of a random string and the softphone’s host name or IP address.”

    Interestingly, the problem is intermittent. Some calls go through. Asterisk must be able to process these calls from time to time. Which is strange on its own.

  6. Bryant Zimmerman said:

    Aug 31, 12 at 8:56 pm

    Vladimir

    We are testing the DP715 very aggressively. We have been please with the units for the most part, but we too have been working bugs with Grandstream. We have several in so far and a number of feature requests as well. I deal directly with several of the support engineers and they bring in the developers when necessary. I would be open to working with you on your issue. If I can create validation tests for your items and reproduce the issue I have had great success getting them to take note and address issues they really do want to address issues. In less than two weeks they have given me test builds address two of our issues and they are working on several others. Because of the cooperation of Grandsteam we are close to being able to offer the DP715 phones to our customers. Even then they will have more items to address to allow for full feature deployments but they are serious about the DP715 product.

    Thanks

    Bryant Zimmerman (ZK Tech Inc.)

    ————————————–

  7. Vladimir Mikhelson said:

    Aug 31, 12 at 9:15 pm

    Bryant,

    Thank you for the reply.

    It looks like either I was very unlucky with the support engineer my SR
    was assigned to or you were extremely lucky. Or maybe Grandstream singles you or your company out for some reason.

    My test is plain vanilla.

    1. Enable SIPS and SRTP for an extension in Asterisk 1.8.15
    2. Sign a certificate on the Asterisk server and provision it manually
    to the DP715
    3. Try calling back and forth.

    My plan was to spend 30 minutes to an hour to test the above and then move to the real-life scenarios. So far I spent 9 days, with no help from Grandstream whatsoever, toying with this test and making no progress.

    The features they must have for real-life deployments:

    * HTTPS on the setup portal with normal set of credentials, i.e. user
    name and password
    * Ability to disable HTTP/HTTPS
    * SSH vs telnet
    * Ability to send host name or other CN not equal to the phone IP in
    TLS negotiation

    I will probably have more after I am past my step 0 testing.

    Thank you, Vladimir

  8. Patrick Lists said:

    Sep 01, 12 at 8:27 am

    [snip]

    Afaik you usually put alternative CNs in SubjectAltName in the certificate. Have you tried that?

    Regards, Patrick

  9. Vladimir Mikhelson said:

    Sep 01, 12 at 10:32 pm

    Patrick,

    Thank you for the hint. I will try and report whether DP715 supported this.

    -Vladimir

  10. Vladimir Mikhelson said:

    Sep 22, 12 at 1:54 am

    Quick update.

    Grandstream finally released the first update to theirDP715 firmware, new v. 1.0.0.8.

    Here are the differences:

    1. I can receive calls over secure SIP and RTP
    2. No outgoing calls go through

    What I observed the phone replies from a different port compared to a port it receives SIP messages on. As a result Asterisk becomes confused. For example, “sip set debug peer 999″ would only track messages to the phone.

    Grandstream’s support is beyond the level of criticism. It takes them
    10 days to reply to a posted message. It seems their only goal is to close the case. So far I am still to see a single bit of help from them.

    I will continue updating this thread.

    -Vladimir

  11. Bryant Zimmerman said:

    Sep 22, 12 at 7:48 am

    Vladimir

    I have been working with Grandstream on the DP715 firmware. Can you give me screen shots of your configs or a download of it, and possibly some asterisk config examples of how your system is set so I can try your configs in our test env. We have the DP715 units working with the new firmware. Also are you dealing with US support or other country? I would like to offer your feed back to the US project engineers for the product, and any info such as ticket numbers and support agent names would be helpful. Your experience of thier support is not the Grandstream I know, and I would like to get to the bottom of the issue.

    Thanks

    Bryant Zimmerman (ZK Tech Inc.)
    616-855-1030 Ext. 2003

    ————————————–

  12. Bryant Zimmerman said:

    Sep 22, 12 at 8:06 am

    Vladimir

    DP715 Phone My primary interest is security. Grandstream claims their intermediate and higher-end models support TLS and SRTP. I am really tired of trying to make Cisco phones to communicate securely with Asterisk. Cisco has a great security model but one has to have their provisioning server for it to function.

    We’ve never had customers ask for this, but if doing so is fairly easy we would look at it as just another feature we push. Do let me know how it works out for you.